similar to: Sine Dialer, GNU dialer, VICIDial and others slightly OT?

I am looking for pros and cons on the Intel Atom cpu. Has anybody been using these in production? I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through except voicemail will be wav format for email purposes(requirement). I will be tying 3 of these together to route

Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

Hello All, I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34 I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been getting: Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136, dlsr=196500 (2:998ms), diff=664 I see an entry in Mantis that Russell fixed code so that this will not show when it shouldn't. Would i be correct in

Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108 sleeping, 0 stopped, 0 zombie Cpu0 : 0.2%us, 0.2%sy, 0.0%ni, 97.3%id, 2.2%wa, 0.1%hi, 0.0%si,

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

hi we need a predictive dialer which can be used with asterisk software. Is it possible? Bye Owais Bin Zuber --------------------------------- Do you Yahoo!? Yahoo! Search - Find what you’re looking for faster. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040310/a49a4b5c/attachment.htm

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

I have a customer with a more or less unique need. Right now we are using Wombat as a dialer software so they can contact clients for QA purposes. Everything is working very well and their contact center productivity is way up from the old manual dialing method. The only thing we are having a problem with is that they have up to 5 phone numbers to contact a single customer. Obviously

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel? Specifically: The predictive dialer dials out over T1 circuits. It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card. In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the

Hi Tilghman!   > Clearly, you missed the point.  Since there is a FREE predictive dialer out > there, and your product costs something, you are not the world's cheapest > predictive dialer.  I respect your wording and the way you or other people think on the list about difference between cheapest and free predictive dialer.   Surely

Currently I have a predictive dialer that is web-enabled, as well as a superdialer mechanism. I was wondering what kind of success people have had with their dialers (lessons learned, etc.) and how I need some direction on how to seamlessly integrate an inbound system with the predictive dialer. My thoughts are that inbound phone calls will be routed to a receptionist first, who will then

Hello everyone, As I am new to the asterisk community (although have been on the list reading for about 6 months) I wanted to see what users would recommend for security to protect several asterisk/ vicidial servers over a fiber connection. Currently I have a managed switch (Tellabs 8813-310) from time warner but I am having intrusion issues on my linux server which I think are contributing

Hello, We've released another update to our Asterisk GUI Client suite: 1.1.7 http://astguiclient.sf.net/ The client suite runs on Windows, UNIX and Mac, includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is

Hello, We've released another update to our Asterisk GUI Client suite: 1.1.9 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in GPL. (the suite

hi wich predicitive dialer are you using and wich one do you recomend? a link to the project/product and a link to a how to will be VERY apreciated. Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi everybody This is Yavuz YILDIRIM I am software developer.I have a some problems in asterisk. I am using mysql db. Realtime using asterisk modules. On db i am using calling hundred fields for use dial. But i don't know how i can automaticly dial this fields on records numbers. Who can help me asterisk api and others. Thank you

Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default